Showing papers on "Telephony published in 2022"

Low Cost VOIP System Incorporation with Raspberry Pi

Abstract: The communication requirements of institutions and corporations have developed over time, and they are using IP based technology moving away from traditional means of voice communication, which are often more expensive. The primary objective of this transition was to increase telephony's effectiveness, adaptability, and cost-effectiveness. Voice Over Internet Protocol (VoIP) is a rising innovation based on contemporary business needs. It's connectivity with the Internet and software implementation have offered this service incredible flexibility when it comes to developing voice communication systems. The Voice Over Internet Protocol (VOIP) is a combination of IP networks, voice apps, and voice calls that has revolutionized the technical or design structure of the phone by replacing the conventional service interaction. VoIP has been widely utilized in business networks as voice and data network convergence provides interoperable services at lower costs. IP-PBX is the PBX telephone switching system that manages internal call switching in an organization . It also allows extensions and external phone lines to communicate with each other.Users using IP phones must register with an asterisk, which is a PBX software implementation that connects users. This system is built around the Raspberry Pi, which is at the core of the concept. To make and receive phone calls, the Session Initiation Protocol (SIP) protocol is employed. Because the Raspberry Pi is programmed at the Linux kernel level, it can work as a server. The system develops codes using free telephony software like as Asterisk and FreePBX, resulting in a cost-effective system.

Problems of Long-Distance Telephony at the Turn of the 19th and 20th Centuries and the Search for Their Solutions

Abstract: At the turn of the 19th and 20th centuries, long-distance telephone communication turned into a serious business, however, the cost of lines was high, and the practical commercial range of telephone signal transmission did not exceed 1600 km. As the lengths of long-distance telephone lines increased, two problems arose that were rec-ognized as particularly relevant, in particular, crosstalk and signal attenuation. AT&T researchers were engaged in solving these problems. This article examines the work of scientists and engineers who have led to real practical solutions to problems of long-distance telephone communication, in particular, Oliver Heaviside, John Stone and George Campbell.

Perancangan Jaringan Voice Over Internet Protokol (Voip) Menggunakan Platform Asterisknow

Abstract: Communication is an important stage to support the running of a work program, without good communication it can cause misunderstandings between the two parties, the importance of the role of communication means several companies are growing rapidly in order to innovate in the manufacture of equipment in the telecommunications sector, which is commonly used in long-distance communication. far is the use of telephone services, both individual telephones and company-based telephones, for some companies that use high-enough telephone activity, this will result in soaring telephone bills that must be paid, thus to reduce the level of communication with high costs, we can use VoIP technology ( Voice Over Internet Protocol) which can be used by utilizing the TCP / IP protocol where the protocol without us realizing it is a protocol used for exchanging data, thus the protocol can also be accessed. used as voice voice data exchange using Open Source Software in the Asterisk application which is used as a substitute for PBX (Private Branch Exchange) devices, where in the application it is used to make the TCP / IP protocol as an intermediary for data exchange to communicate or Voice. The advantage of this application is that it is able to serve a maximum of 1000 online accounts on one server and 240 concurrent calls. This is of course suitable for companies who want to implement this technology based on open source applications.

Subjective Quality Assessment of One-to-One Video-Telephony Services

Abstract: Video-telephony applications have been widely used in people's daily life, such as online conferences, online education, and socialization. Especially during the COVID-19 pandemic, the business volume of video-telephony services has generally increased rapidly. This leads to a growing need for video-telephony service quality assessment and monitoring. This paper presents the subjective tests conducted in 'Computational model used as a QoE/QoS monitor to assess video-telephony services' (G.CMVTQS) project, which is under study in ITU-T SG12 Q.15. Two types of subjective tests are designed. Audiovisual material subjective test focuses on the quality of multimedia streams, while conversational subjective test focuses on the quality of interaction experience. These tests simulate real-life situations in video-telephony with different types of user terminals (PCs, TVs and mobile phones), environments (office, home, restaurant and outdoor), audio/video parameters (resolution, frame rate, bit rate and codec) and network impairments (packet loss, delay, jitter, bandwidth, synchronization). The final subjective database built collaboratively by five laboratories in different countries will be analyzed in the near future and provide more insights to develop new QoE/QoS monitor models for video-telephony.

Is Having an Expert “Friend” Enough? An Analysis of Consumer Switching Behavior in Mobile Telephony

Abstract: We present novel evidence from a large panel of UK consumers who receive personalized reminders from a specialist price-comparison website about the precise amount they could save by switching to their best-suited alternative mobile telephony plan. We document three phenomena. First, even selfregistered consumers with positive savings exhibit inertia. Second, we show that being informed about potential savings has a positive and significant effect on switching. Third, controlling for savings, the effect of incurring overage payments is significant and similar in magnitude to the effect of savings: paying an amount that exceeds the recurrent monthly fee weighs more on the switching decision than being informed that one can save that same amount by switching to a less inclusive plan. We interpret this asymmetric reaction on switching behavior as potential evidence of loss aversion. In other words, when facing complex and recurrent tariff plan choices, consumers care about savings but also seem to be willing to pay upfront fees in order to get “peace of mind”. Reference Details CWPE 2351 Published 11 July 2023

Telephone Systems

Abstract: While the tried and tested telephone networkTelephone networks is gradually being replaced by the new VoIP technology, it is unfortunately very rarely made clear that the use of so-called InternetInternet telephony also entails massive risks. For this reason, VoIP telephone systemsTelephone systems are often operated without adequate protection. This chapter explains, using concrete examples, what the dangers of using VoIP telephone systemsTelephone systems can look like and how one should protect oneself from the corresponding dangers.

Speech Enhancement Techniques in the Digital, Analog, and Hybrid Domains

Abstract: Speech intelligibility, clarity, and naturalness may all be improved by the use of signal processing techniques. You may come across similar processing in applications like mobile telephony, internet telephony, or the personal intercom. It is possible to increase speech quality or performance by using implementation-specific benefits via sophisticated domain selection for implementation (digital, analogue, and hybrid). Speech-enhancement applications in the three domains of implementation are the focus of this research. Using a simple but effective speech enhancement method, the advantages and disadvantages of each implementation will be shown

Virtual Asterisk IP-PBX Operation Studying and Exploring at the University

Abstract: A brief outline of the ITU-T concept of fixed mobile Future Networks is given. According to the concept, Future Networks are to support the virtualization of network resources. Studying the principles of virtualizing telecommunications systems and communications networks is a part of Infocomm Technologies and Communications Networks educational programme at the Moscow Technical University of Communications and Informatics. The students are taught how to install, set up and operate a virtual Asterisk IP-PBX. They study such additional features as call transfer and pickup, creating an interactive voice response, videoconferences. They also explore the interaction of the Asterisk IP-PBX and the public switched telephone network as well as the quality of service in IP telephony.

Low Cost Implementation of Multiple Services in a Telephone Plant Hosted in a Cloud Infrastructure

Abstract: This paper present the implementation of an OpenSource telephone plant called Issabel, a Linux based PBX (private telephone network), totally hosted in a cloud infrastructure, with the aim of taking advantage of all its features and functionalities such as email server, chat server, call center, etc. Allowing agents to connect from anywhere in the world and allowing cost reduction compared to traditional implementations, importance is given to the cybersecurity aspects involved in the internet. The connection with the public switched telephone network is investigated and implemented. Finally, it is made a cost comparison between implementing cloud services and how it was done traditionally.

When Evil Calls: Targeted Adversarial Voice over IP Network

Abstract: As the COVID-19 pandemic fundamentally reshaped the remote life and working styles, Voice over IP (VoIP) telephony and video conferencing have become a primary method of connecting communities together. However, little has been done to understand the feasibility and limitations of delivering adversarial voice samples via such communication channels. In this paper, we propose TAINT Targeted Adversarial Voice over IP Network, the first targeted, query-efficient, hard label blackbox, adversarial attack on commercial speech recognition platforms over VoIP. The unique channel characteristics of VoIP pose significant new challenges, such as signal degradation, random channel noise, frequency selectivity, etc. To address these challenges, we systematically analyze the structure and channel characteristics of VoIP through reverse engineering. A noise-resilient efficient gradient estimation method is then developed to ensure a steady and fast convergence of the adversarial sample generation process. We demonstrate our attack in both over-the-air and over-the-line settings on four commercial automatic speech recognition (ASR) systems over the five most popular VoIP Conferencing Software (VCS). We show that TAINT can achieve performance that is comparable to the existing methods even with the addition of VoIP channel. Even in the most challenging scenario where there is an active speaker in Zoom, TAINT can still succeed within 10 attempts while staying out of the speaker focus of the video conference1. 1Attack demos against Google Assistant, Amazon Echo, Microsoft Cortana, as well as adversarial audio samples of different lengths and source code of the project are available on the website: https://sites.google.com/view/targeted-adversarial-voip. Permission to make digital or hard copies of part or all of this work for personal or classroom use is granted without fee provided that copies are not made or distributed for profit or commercial advantage and that copies bear this notice and the full citation on the first page. Copyrights for third-party components of this work must be honored. For all other uses, contact the owner/author(s). CCS ’22, November 7–11, 2022, Los Angeles, CA, USA © 2022 Copyright held by the owner/author(s). ACM ISBN 978-1-4503-9450-5/22/11. https://doi.org/10.1145/3548606.3560671 CCS CONCEPTS • Security and privacy→ Software and application security; • Computing methodologies→Machine learning.

The Norwegian Mobile Telephony and Internet Markets

Abstract: Abstract Norway has been a pioneer in the development and adoption of the Internet and mobile telephony technologies. Already from an early stage, Norway was involved in research, development, and testing of initiatives such as the ARPANET project and the Nordic Mobile Telephone (NMT) system. Today, access to the Internet and mobile technologies—including smartphones—is globally widespread. The major objective of this chapter is to describe how the market for the Internet and mobile telephony in Norway has evolved since its inception in the 1970s until today. The historical and current market structure of telecommunications is discussed. Moreover, the chapter investigates the role and significance of mobile virtual network operators (MVNOs). Finally, the chapter examines the regulations imposed by the Norwegian Communications Authority (Nkom) on dominant stakeholders in the Norwegian telecom market.

A Study of Effect of SPQ on IP Telephony Service with Binaural 3D Sounds

Abstract: Since QoE of an IP telephony service with binaural 3D sounds is affected its QoS, it is necessary to control QoS to improve the QoE. This paper adopts Strict Priority Queueing (SPQ) defined in IEEE 802.1TSN(Time-Sensitive Networking) as such a QoS control. This study also evaluates the QoE provided by the IP telephony service with binaural 3D sounds over SPQ-enabled networks by experiments with subjective evaluation. The experimental results show that SPQ can improve QoE of the IP telephony service with binaural 3D sounds.

Speaker Recognition on Mono-Channel Telephony Recordings

Abstract: Conversations stored as mono data is a common problem in many real world speaker recognition applications. In this paper, we focus on investigative scenarios, where a number of mono telephone conversations are available for a speaker of interest. For example, a human operator may have verified that the speaker is present in these conversations. We propose several approaches for automatically creating enrollment models for the speaker of interest from such data. We then use the enrollment models to search for appearances of the speaker of interest in other calls. We analyze the performance of the different method on two dataset that matches our scenario, one is from a simulated case and one is from a real case. and real databases. We show that even simple methods not requiring tunable settings can perform well in these challenging and unpredicted scenarios. Nevertheless, bigger databases should be used to confirm these findings. The meth-198

Observing the Performance of Interface Device for Communicating Based on a Half-Duplex System by Giving a Realistic Condition

Abstract: This article explains about a stage after the stage of the verification test, there is a phase of a form of prior activity to the final stage, i.e., the hot commissioning phase. Hot commissioning is a stage for providing a realistic condition or validation test activity. Based on the follow-up activity which is the reason for choosing the title, therefore the two research objectives in this article are set. The first objective is to create conditions for the system when the end user controls through the radio-based electronic device, whereas the second objective is to create conditions for the system when the end user controls through the telephony-based electronic device. The steps of research methods for achieving the research objectives are carried out through (i) calling activities, (ii) connecting process, and (iii) disconnecting activities after obtaining a connection to communicate from a radio-based device (using the handy talky, HT) and from the telephony-based device (used the mobile phone). The two results based on research objectives have been obtained after observation regarding activities of calling activity, making telephone calls, and termination activity after connecting from radio-based and telephony-based devices. Based on the research results, it can be concluded that the half duplex-based interface device for communicating (IDC) functioned after the realistic conditions is given.

Michael Pupin and Pupinization of Telephone Lines

Abstract: The scientific ways that led the American radio engineer Michael Pupin to the development of telephone technologies aimed at improving the quality of an audio signal when it is transmitted over long distances are considered. Pupin’s important inventions in the field of long-distance telephony are investigated. His theories of using inductors to reduce the attenuation of the transmitted telegraph and telephone signal over the cable by artificially increasing its inductance are described. Attention is paid to the dispute between Pupin and Campbell in the primacy of the invention of load coils and its most significant consequences.

Data over Telephony Line

Abstract: The pulse code modulation process was applied at the operator to convert the analog waveform to a digital 64 kbps signal transported over digital trunks. Since digital subscriber line technologies rely on the bandwidth of subscriber lines being higher than that of the rest of the telephone network, the data had to be separated from the voice signals at the end of the subscriber lines. Asymmetric Digital Subscriber Line (ADSL) uses two ways of transporting the data depending on the Quality of Service requirements of the data streams. This chapter describes the initialization sequence of an ADSL line as per T.413 and G.992.1 specifications and an ADSL2 line as per G.992.3. It also discusses the cold start initialization process of a very high-speed digital subscriber line (VDSL) as per ITU-T G.993.1 and VDSL2 line as per G.993.2 specifications.

Performance Assessment of Simulink Based Speech Radio Band Extension Technique on Elderly People

Abstract: The main drawback of conventional communication networks is the limitation of the frequency band. Proper speech frequency for telecommunication systems spans from 3 to 3.4 Khz and is popularly known as In-Band (IB). The IB does have a significant impact on communication apprehension and interpretation. It has a more significant impact on senior adults over 60. The Telephony Speech Spectrum Enhancement (TSSE) Technique, when used in conjunction with the IB speech signal, tries to improve the IB speech signal's performance and perception by transmitting erroneous audio frequency information as additional data. On The receiver end, these omitted voice spectral characteristics sent as extra data help build an improved Wide Band (WB) signal. This work simultaneously builds the transmitter and the receiver with Simulink and investigates the potential of TSSE to boost the perception of telephonic conversation among older adults. During this investigation, the effectiveness of the TSSE approach is analysed by contrasting it with a Conventional Telephony Signal Enhancement Technique (CTSET). Subjective as well as quantitative tests analysed and validate the improved performance in the grade and perception of IB signal for the older adult population.

Efficient Use of Large Pre-Trained Models for Low Resource ASR

Abstract: Automatic speech recognition (ASR) has been established as a well-performing technique for many scenarios where lots of labeled data is available. Additionally, unsupervised representation learning recently helped to tackle tasks with limited data. Following this, hardware limitations and applications give rise to the question how to efficiently take advantage of large pretrained models and reduce their complexity for downstream tasks. In this work, we study a challenging low resource conversational telephony speech corpus from the medical domain in Vietnamese and German. We show the benefits of using unsupervised techniques beyond simple fine-tuning of large pre-trained models, discuss how to adapt them to a practical telephony task including bandwidth transfer and investigate different data conditions for pre-training and fine-tuning. We outperform the project baselines by 22% relative using pretraining techniques. Further gains of 29% can be achieved by refinements of architecture and training and 6% by adding 0.8 h of in-domain adaptation data.

Analytical modeling and performance evaluation of SIP signaling protocol: Analytical modeling of SIP

Abstract: In the past few years, Voice over Internet Protocol (VoIP) has emerged as the most preferred method of making voice calls over the Internet (Internet telephony) because of its low service cost and value-added features. Furthermore, Session Initiation Protocol (SIP) is the most recent application-layer protocol in the Transmission Control Protocol/Internet Protocol suite, widely used in Internet telephony (VoIP). It is a protocol of mounting importance, and hence it is highly desirable to study and evaluate its performance measures. In this chapter, we propose an analytical model of SIP under an unreliable transport protocol by using stochastic reward net (SRN) modeling technique. The objective of developing the analytical model for SIP is the performance evaluation of the SIP INVITE request. This chapter focuses on the INVITE request flow, which initiates a session between the client and the server. The proposed model incorporates all the important SIP features such as INVITE request, different types of responses, various timers, acknowledgment generation, retransmissions, and transport errors. In particular, certain measures of performance of SIP INVITE request such as throughput and latency are computed through the proposed SRN model. The viability of the proposed model is exhibited through numerical illustration. The suggested analytical model is also supported via simulation through MATLAB.

Telemarketing automation based on the MIKO IP-telephony module

Abstract: In modern conditions, an integral part of the functioning of business processes and the company as a whole is the information system. The information system allows you to significantly speed up and automate the execution of most of the processes in the company. The main value of business process automation is the ability to register various kinds of information, thanks to which an automated information system will generate most of the documentation automatically. The work presents the integration of the 1C program and IP-telephony. Automation eliminates unnecessary actions of the manager when searching for a conversation record on the PBX server. To listen to the recording of the conversation, the form "processing applications" contains all the necessary data. As a result, automation helped to avoid wasting a lot of time duplicating a phone number on a hardware phone, as well as making mistakes in dialing. The IS, converting the phone number indicated in the document into the desired format, makes a call to the client.

Solutions for Closing Usage Gap in Rural Areas in West Africa

Abstract: Bridging the digital gap in general, and the rural usage gap in particular, has been a significant concern for the telephony industry for years. This study examines the key factors in telephony adoption by non-subscriber in West Africa. To do this, we used a quantitative approach using the structural equation model. We collected data in two representative West African countries: Nigeria and Guinea Conakry. On a global population of 620 people, 204 did not own a mobile phone. Analysis of the data collected from this 204 non-subscriber showed that the key factors driving future technology adoption are attitude, social influence, perceived usefulness, and perceived ease of use the demographic factor of gender moderates these factors.

Implementation of voice call transfer service between smart phone and tablet through Wi-Fi

Abstract: Communication through voice call leads to significant growth in technology in distant areas where two or more people from opposite ends of world will connect. This research describes a case study of voice call transfer service. This research aims at designing a system that will allow Android users to communicate over Wi-Fi. This design is able to transfer voice of incoming telephone caller over Wi-Fi network at real time through UDP. It uses client/server architecture: Server for receiving telephone call and transferring voice (one user) and client for receiving incoming caller voice and enables communication with server. Architecture designed could be used on Android smart phones with telephony enabled and tablets with telephony not enabled. Outcome of this research will allow users to communicate on real time at no cost. Proposed design gives cost effective, reliable and real time voice communication over Wi-Fi. It provides good and comfort experience to users in emergency situation where user cannot effort cost for telephone call. Proposed design is useful for educational organizations, construction buildings, shopping malls and hospitals which point to new possibilities for voice communication.

Method for ensuring information security of a VOIP telephony network with a forecast of an intruder's intrusion strategy

Abstract: Introduction: the development of technologies in the field of information and telecommunications, as well as the unification of networks, and in particular the construction of distributed VoIP telephony networks, allow us to formulate the problem that the known methods of managing the protection of VoIP networks are not effective enough in modern conditions, since they take into account only one side of the information confrontation. Purpose: To develop a method for ensuring the information security of a VoIP telephony network, which allows to increase the probability of VoIP network security by reducing the time required for analyzing the actions of the violator, analyzing and processing risks under the influence of the violator. Results: Based on the proposed structure of an information security management system integrated into a VoIP network, a method for ensuring the information security of a VoIP telephony network under the influence of an intruder has been developed by introducing decision-making support processes in the VoIP network information security management system using intelligent intrusion detection tools distributed across segments. This method allows you to build a graph of events of the intruder's actions, on the basis of which mathematical modeling of MiTM and SPIT attacks on the VoIP telephony network is carried out. As a result of the simulation, the dependence of the successful impact on the internal and external characteristics of attacks is obtained, which is the main one of the developed software, which allows to obtain the values of the probability of security of the VoIP network from the parameters of the intruder's impact for further selection of adequate measures for managing the information security of the VoIP telephony network. The method includes the processes of analyzing the digital stream and determining the parameters of protocols and profiles of intruder attacks. Practical relevance: The developed method provides an opportunity to study issues aimed at the security of the VoIP-telephony network, which is affected by violators.

Fixed Telephone Networks

Abstract: The first digital voice encoding method, pulse code modulation (PCM), had a profound effect on the design of the telephony network and transmission systems for which the basic bitrate 64 kbps equals the PCM codec bitrate. The data rate of PCM coded signal is constant 64 kbps, as the sampling produces 8-bit code words for each of the 8000 samples taken in a second. This data rate is the basic rate of all digital fixed telephone networks. Although the exchange designs are vendor specific, this chapter describes a typical architecture of a digital telephone exchange. Synchronous Digital Hierarchy (SDH) provides a way to multiplex many E1, T1, or plesiochronous digital hierarchy signals synchronously to one single electric or optical cable without additional justification bits. The chapter also provides high-level architectural overview of SDH. Wavelength division multiplexing was the emerging solution to support multiplexing of different optical wavelengths to one optical cable.